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ppsspp/ext/ffmpeg/libavcodec/aacsbr.c
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Henrik RydgårdandClaude Opus 5.5 d0e3141f43 FFmpeg: Vendor what PPSSPP uses into ext/ffmpeg, drop the submodule 
The ffmpeg submodule held prebuilt libraries for every platform next to the
sources (about 1.5 GB checked out), and nothing uses those any more. Only the
needed sources are copied in, from the submodule's commit, along with every
header and template they include. The path is new, so checking out commits on
either side of this works.

It's also cut down to what PPSSPP uses: the H.264, AAC, MP3, ATRAC3 and ATRAC3+
decoders, the H.264 parser, the demuxers as before, and the FFV1 encoder, AVI
muxer and file protocol for AVI dumping. config.h takes its CONFIG_ values from
FFmpeg 3.0.2's own configure run with just those enabled. The raw H.264 demuxer
stays even though nothing opens raw H.264: probing PSMF data uses it to identify
the video stream's codec, which otherwise comes out as none.

On top of that:
- The files nothing links (configure would build them, but they're never pulled
  out of the static library, going by the GCC link map) are left out: mostly
  libavutil's crypto and hashing and the alternative DCT/IDCT/FFT versions.
- libswresample isn't built, now that nothing uses it.
- The H.264 decoder only does 8-bit video, which is all the PSP has. The SPS
  parser rejects any other bit depth, and the DSP files no longer build their 9-
  to 16-bit variants, which the linker kept because the init functions pick one
  by bit depth at run time.

That's 160 files to compile instead of 277, and 391 of FFmpeg's vendored, about
5.8 MB. The README in ext/ffmpeg has what was done, the PPSSPP patches in the
copy, how to add a component, and the tradeoffs.

Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
2026-10-25 22:27:54 +01:00

370 lines
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/*
* AAC Spectral Band Replication decoding functions
* Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
* Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* AAC Spectral Band Replication decoding functions
* @author Robert Swain ( rob opendot cl )
*/
#define USE_FIXED 0
#include "aac.h"
#include "sbr.h"
#include "aacsbr.h"
#include "aacsbrdata.h"
#include "aacsbr_tablegen.h"
#include "fft.h"
#include "internal.h"
#include "aacps.h"
#include "sbrdsp.h"
#include "libavutil/internal.h"
#include "libavutil/libm.h"
#include "libavutil/avassert.h"
#include <stdint.h>
#include <float.h>
#include <math.h>
#if ARCH_MIPS
#include "mips/aacsbr_mips.h"
#endif /* ARCH_MIPS */
static VLC vlc_sbr[10];
static void aacsbr_func_ptr_init(AACSBRContext *c);
static void make_bands(int16_t* bands, int start, int stop, int num_bands)
{
int k, previous, present;
float base, prod;
base = powf((float)stop / start, 1.0f / num_bands);
prod = start;
previous = start;
for (k = 0; k < num_bands-1; k++) {
prod *= base;
present = lrintf(prod);
bands[k] = present - previous;
previous = present;
}
bands[num_bands-1] = stop - previous;
}
/// Dequantization and stereo decoding (14496-3 sp04 p203)
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
{
int k, e;
int ch;
static const double exp2_tab[2] = {1, M_SQRT2};
if (id_aac == TYPE_CPE && sbr->bs_coupling) {
int pan_offset = sbr->data[0].bs_amp_res ? 12 : 24;
for (e = 1; e <= sbr->data[0].bs_num_env; e++) {
for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
float temp1, temp2, fac;
if (sbr->data[0].bs_amp_res) {
temp1 = ff_exp2fi(sbr->data[0].env_facs_q[e][k] + 7);
temp2 = ff_exp2fi(pan_offset - sbr->data[1].env_facs_q[e][k]);
}
else {
temp1 = ff_exp2fi((sbr->data[0].env_facs_q[e][k]>>1) + 7) *
exp2_tab[sbr->data[0].env_facs_q[e][k] & 1];
temp2 = ff_exp2fi((pan_offset - sbr->data[1].env_facs_q[e][k])>>1) *
exp2_tab[(pan_offset - sbr->data[1].env_facs_q[e][k]) & 1];
}
if (temp1 > 1E20) {
av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
temp1 = 1;
}
fac = temp1 / (1.0f + temp2);
sbr->data[0].env_facs[e][k] = fac;
sbr->data[1].env_facs[e][k] = fac * temp2;
}
}
for (e = 1; e <= sbr->data[0].bs_num_noise; e++) {
for (k = 0; k < sbr->n_q; k++) {
float temp1 = ff_exp2fi(NOISE_FLOOR_OFFSET - sbr->data[0].noise_facs_q[e][k] + 1);
float temp2 = ff_exp2fi(12 - sbr->data[1].noise_facs_q[e][k]);
float fac;
av_assert0(temp1 <= 1E20);
fac = temp1 / (1.0f + temp2);
sbr->data[0].noise_facs[e][k] = fac;
sbr->data[1].noise_facs[e][k] = fac * temp2;
}
}
} else { // SCE or one non-coupled CPE
for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
if (sbr->data[ch].bs_amp_res)
sbr->data[ch].env_facs[e][k] = ff_exp2fi(sbr->data[ch].env_facs_q[e][k] + 6);
else
sbr->data[ch].env_facs[e][k] = ff_exp2fi((sbr->data[ch].env_facs_q[e][k]>>1) + 6)
* exp2_tab[sbr->data[ch].env_facs_q[e][k] & 1];
if (sbr->data[ch].env_facs[e][k] > 1E20) {
av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
sbr->data[ch].env_facs[e][k] = 1;
}
}
for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
for (k = 0; k < sbr->n_q; k++)
sbr->data[ch].noise_facs[e][k] =
ff_exp2fi(NOISE_FLOOR_OFFSET - sbr->data[ch].noise_facs_q[e][k]);
}
}
}
/** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
* (14496-3 sp04 p214)
* Warning: This routine does not seem numerically stable.
*/
static void sbr_hf_inverse_filter(SBRDSPContext *dsp,
float (*alpha0)[2], float (*alpha1)[2],
const float X_low[32][40][2], int k0)
{
int k;
for (k = 0; k < k0; k++) {
LOCAL_ALIGNED_16(float, phi, [3], [2][2]);
float dk;
dsp->autocorrelate(X_low[k], phi);
dk = phi[2][1][0] * phi[1][0][0] -
(phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
if (!dk) {
alpha1[k][0] = 0;
alpha1[k][1] = 0;
} else {
float temp_real, temp_im;
temp_real = phi[0][0][0] * phi[1][1][0] -
phi[0][0][1] * phi[1][1][1] -
phi[0][1][0] * phi[1][0][0];
temp_im = phi[0][0][0] * phi[1][1][1] +
phi[0][0][1] * phi[1][1][0] -
phi[0][1][1] * phi[1][0][0];
alpha1[k][0] = temp_real / dk;
alpha1[k][1] = temp_im / dk;
}
if (!phi[1][0][0]) {
alpha0[k][0] = 0;
alpha0[k][1] = 0;
} else {
float temp_real, temp_im;
temp_real = phi[0][0][0] + alpha1[k][0] * phi[1][1][0] +
alpha1[k][1] * phi[1][1][1];
temp_im = phi[0][0][1] + alpha1[k][1] * phi[1][1][0] -
alpha1[k][0] * phi[1][1][1];
alpha0[k][0] = -temp_real / phi[1][0][0];
alpha0[k][1] = -temp_im / phi[1][0][0];
}
if (alpha1[k][0] * alpha1[k][0] + alpha1[k][1] * alpha1[k][1] >= 16.0f ||
alpha0[k][0] * alpha0[k][0] + alpha0[k][1] * alpha0[k][1] >= 16.0f) {
alpha1[k][0] = 0;
alpha1[k][1] = 0;
alpha0[k][0] = 0;
alpha0[k][1] = 0;
}
}
}
/// Chirp Factors (14496-3 sp04 p214)
static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
{
int i;
float new_bw;
static const float bw_tab[] = { 0.0f, 0.75f, 0.9f, 0.98f };
for (i = 0; i < sbr->n_q; i++) {
if (ch_data->bs_invf_mode[0][i] + ch_data->bs_invf_mode[1][i] == 1) {
new_bw = 0.6f;
} else
new_bw = bw_tab[ch_data->bs_invf_mode[0][i]];
if (new_bw < ch_data->bw_array[i]) {
new_bw = 0.75f * new_bw + 0.25f * ch_data->bw_array[i];
} else
new_bw = 0.90625f * new_bw + 0.09375f * ch_data->bw_array[i];
ch_data->bw_array[i] = new_bw < 0.015625f ? 0.0f : new_bw;
}
}
/**
* Calculation of levels of additional HF signal components (14496-3 sp04 p219)
* and Calculation of gain (14496-3 sp04 p219)
*/
static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr,
SBRData *ch_data, const int e_a[2])
{
int e, k, m;
// max gain limits : -3dB, 0dB, 3dB, inf dB (limiter off)
static const float limgain[4] = { 0.70795, 1.0, 1.41254, 10000000000 };
for (e = 0; e < ch_data->bs_num_env; e++) {
int delta = !((e == e_a[1]) || (e == e_a[0]));
for (k = 0; k < sbr->n_lim; k++) {
float gain_boost, gain_max;
float sum[2] = { 0.0f, 0.0f };
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
const float temp = sbr->e_origmapped[e][m] / (1.0f + sbr->q_mapped[e][m]);
sbr->q_m[e][m] = sqrtf(temp * sbr->q_mapped[e][m]);
sbr->s_m[e][m] = sqrtf(temp * ch_data->s_indexmapped[e + 1][m]);
if (!sbr->s_mapped[e][m]) {
sbr->gain[e][m] = sqrtf(sbr->e_origmapped[e][m] /
((1.0f + sbr->e_curr[e][m]) *
(1.0f + sbr->q_mapped[e][m] * delta)));
} else {
sbr->gain[e][m] = sqrtf(sbr->e_origmapped[e][m] * sbr->q_mapped[e][m] /
((1.0f + sbr->e_curr[e][m]) *
(1.0f + sbr->q_mapped[e][m])));
}
}
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
sum[0] += sbr->e_origmapped[e][m];
sum[1] += sbr->e_curr[e][m];
}
gain_max = limgain[sbr->bs_limiter_gains] * sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
gain_max = FFMIN(100000.f, gain_max);
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
float q_m_max = sbr->q_m[e][m] * gain_max / sbr->gain[e][m];
sbr->q_m[e][m] = FFMIN(sbr->q_m[e][m], q_m_max);
sbr->gain[e][m] = FFMIN(sbr->gain[e][m], gain_max);
}
sum[0] = sum[1] = 0.0f;
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
sum[0] += sbr->e_origmapped[e][m];
sum[1] += sbr->e_curr[e][m] * sbr->gain[e][m] * sbr->gain[e][m]
+ sbr->s_m[e][m] * sbr->s_m[e][m]
+ (delta && !sbr->s_m[e][m]) * sbr->q_m[e][m] * sbr->q_m[e][m];
}
gain_boost = sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
gain_boost = FFMIN(1.584893192f, gain_boost);
for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
sbr->gain[e][m] *= gain_boost;
sbr->q_m[e][m] *= gain_boost;
sbr->s_m[e][m] *= gain_boost;
}
}
}
}
/// Assembling HF Signals (14496-3 sp04 p220)
static void sbr_hf_assemble(float Y1[38][64][2],
const float X_high[64][40][2],
SpectralBandReplication *sbr, SBRData *ch_data,
const int e_a[2])
{
int e, i, j, m;
const int h_SL = 4 * !sbr->bs_smoothing_mode;
const int kx = sbr->kx[1];
const int m_max = sbr->m[1];
static const float h_smooth[5] = {
0.33333333333333,
0.30150283239582,
0.21816949906249,
0.11516383427084,
0.03183050093751,
};
float (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
int indexnoise = ch_data->f_indexnoise;
int indexsine = ch_data->f_indexsine;
if (sbr->reset) {
for (i = 0; i < h_SL; i++) {
memcpy(g_temp[i + 2*ch_data->t_env[0]], sbr->gain[0], m_max * sizeof(sbr->gain[0][0]));
memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0], m_max * sizeof(sbr->q_m[0][0]));
}
} else if (h_SL) {
for (i = 0; i < 4; i++) {
memcpy(g_temp[i + 2 * ch_data->t_env[0]],
g_temp[i + 2 * ch_data->t_env_num_env_old],
sizeof(g_temp[0]));
memcpy(q_temp[i + 2 * ch_data->t_env[0]],
q_temp[i + 2 * ch_data->t_env_num_env_old],
sizeof(q_temp[0]));
}
}
for (e = 0; e < ch_data->bs_num_env; e++) {
for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
memcpy(g_temp[h_SL + i], sbr->gain[e], m_max * sizeof(sbr->gain[0][0]));
memcpy(q_temp[h_SL + i], sbr->q_m[e], m_max * sizeof(sbr->q_m[0][0]));
}
}
for (e = 0; e < ch_data->bs_num_env; e++) {
for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
LOCAL_ALIGNED_16(float, g_filt_tab, [48]);
LOCAL_ALIGNED_16(float, q_filt_tab, [48]);
float *g_filt, *q_filt;
if (h_SL && e != e_a[0] && e != e_a[1]) {
g_filt = g_filt_tab;
q_filt = q_filt_tab;
for (m = 0; m < m_max; m++) {
const int idx1 = i + h_SL;
g_filt[m] = 0.0f;
q_filt[m] = 0.0f;
for (j = 0; j <= h_SL; j++) {
g_filt[m] += g_temp[idx1 - j][m] * h_smooth[j];
q_filt[m] += q_temp[idx1 - j][m] * h_smooth[j];
}
}
} else {
g_filt = g_temp[i + h_SL];
q_filt = q_temp[i];
}
sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
i + ENVELOPE_ADJUSTMENT_OFFSET);
if (e != e_a[0] && e != e_a[1]) {
sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
q_filt, indexnoise,
kx, m_max);
} else {
int idx = indexsine&1;
int A = (1-((indexsine+(kx & 1))&2));
int B = (A^(-idx)) + idx;
float *out = &Y1[i][kx][idx];
float *in = sbr->s_m[e];
for (m = 0; m+1 < m_max; m+=2) {
out[2*m ] += in[m ] * A;
out[2*m+2] += in[m+1] * B;
}
if(m_max&1)
out[2*m ] += in[m ] * A;
}
indexnoise = (indexnoise + m_max) & 0x1ff;
indexsine = (indexsine + 1) & 3;
}
}
ch_data->f_indexnoise = indexnoise;
ch_data->f_indexsine = indexsine;
}
#include "aacsbr_template.c"
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